avconv settings for replacing ffmpeg in Ubuntu
Posted: Fri Nov 02, 2012 3:49 pmI have not been following the issues with ffmpeg and avconv forking, but the ubuntu build seems to have deprecated ffmpeg in favor of avconv. So I decided to post the settings I am using for avconv in case anyone is having conversion issues.
Although the avconv syntax is almost the same as ffmpeg there are few changes (changed bitrate from -ab to -b, the quality from -v to -q. The default output for avconv seemed pretty verbose, so I added -loglevel error):
New Subsonic Settings:
Name: mp3 audio
Convert From: ogg oga aac m4a flac wav wma aif aiff ape mpc shn
Convert To: mp3
Step 1: avconv -i %s -b %bk -q 0 -loglevel error -f mp3 -
Step 2 is blank since its not needed.
Here is an explanation of the switches to help anyone understand:
-i %s (take the input from the file stream %s)
-b %bk (set the bitrate to the variable %b and add a k since the bitrate is in thousands, ie "128k")
-q 0 (set the quality to the best, this value varies from 0 to 31, with 31 being the worst. Could lower if your machine processor sucked)
-loglevel error (set the output logging level to errors only)
-f mp3 (set the output format to mp3)
- (direct the output to the pipe so that it can be streamed to the player)
NOTE: I noticed m4a transcoding was not working and was giving the error "Encoder (coded id 0) not found for output stream #0:0" error message in the log. This was due to the upgraded Ubuntu not having all the transcoders by default (because they are restricted by law without a license). In order to install those transcoders, one should install the "ubuntu-restricted-extras" package in the Ubuntu Software Center. This will load transcoders needed to do many of the audio format conversions you might have.
Although the avconv syntax is almost the same as ffmpeg there are few changes (changed bitrate from -ab to -b, the quality from -v to -q. The default output for avconv seemed pretty verbose, so I added -loglevel error):
New Subsonic Settings:
Name: mp3 audio
Convert From: ogg oga aac m4a flac wav wma aif aiff ape mpc shn
Convert To: mp3
Step 1: avconv -i %s -b %bk -q 0 -loglevel error -f mp3 -
Step 2 is blank since its not needed.
Here is an explanation of the switches to help anyone understand:
-i %s (take the input from the file stream %s)
-b %bk (set the bitrate to the variable %b and add a k since the bitrate is in thousands, ie "128k")
-q 0 (set the quality to the best, this value varies from 0 to 31, with 31 being the worst. Could lower if your machine processor sucked)
-loglevel error (set the output logging level to errors only)
-f mp3 (set the output format to mp3)
- (direct the output to the pipe so that it can be streamed to the player)
NOTE: I noticed m4a transcoding was not working and was giving the error "Encoder (coded id 0) not found for output stream #0:0" error message in the log. This was due to the upgraded Ubuntu not having all the transcoders by default (because they are restricted by law without a license). In order to install those transcoders, one should install the "ubuntu-restricted-extras" package in the Ubuntu Software Center. This will load transcoders needed to do many of the audio format conversions you might have.